EP0689191A2 - Emetteur-récepteur mobile - Google Patents

Emetteur-récepteur mobile Download PDF

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Publication number
EP0689191A2
EP0689191A2 EP95201578A EP95201578A EP0689191A2 EP 0689191 A2 EP0689191 A2 EP 0689191A2 EP 95201578 A EP95201578 A EP 95201578A EP 95201578 A EP95201578 A EP 95201578A EP 0689191 A2 EP0689191 A2 EP 0689191A2
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EP
European Patent Office
Prior art keywords
values
speech
delay
speech signal
signal
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EP95201578A
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German (de)
English (en)
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EP0689191A3 (fr
EP0689191B1 (fr
Inventor
Martin Dipl.-Ing. C/O Philips Rainer
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Philips Intellectual Property and Standards GmbH
Koninklijke Philips NV
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Philips Corporate Intellectual Property GmbH
Philips Patentverwaltung GmbH
Koninklijke Philips Electronics NV
Philips Electronics NV
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Publication of EP0689191A3 publication Critical patent/EP0689191A3/fr
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02166—Microphone arrays; Beamforming

Definitions

  • the invention relates to a mobile radio terminal with a speech processing device.
  • noise signals are often contained in speech signals to be processed, which leads to a reduction in speech quality and thus, in particular, to poor speech intelligibility.
  • This problem occurs, for example, in the case of mobile radio terminals which are used in motor vehicles and have a hands-free device.
  • Speech signals that are received by microphones of the hands-free system arranged in the motor vehicle contain, on the one hand, speech signal components that are generated by the respective user (voice source) of the mobile radio terminal within the motor vehicle, and, on the other hand, noise signal components that result from other ambient noises and while driving, essentially from the engine and Driving noises exist.
  • the difference between two samples of two mutually time-shifted signals is formed to form the error values, one of the signals being delayed.
  • the corresponding delay value is rounded to an integer multiple of a sampling interval of the signals. Convergence problems occur in such a way that strong oscillations of the rounded delay values occur when very small error values are reached.
  • the delay values oscillate between two rounded delay values at a sampling interval.
  • the invention has for its object to improve the speech quality of the speech signals to be processed and to reduce convergence problems.
  • the gradient estimates serve to estimate the respective gradient of the performance of the error values or, in other words, the squared error values.
  • the control means determine the delay estimates so that the performance of the error values is reduced.
  • the convergence of the delay values determined from the delay estimated values is considerably improved since the delay estimated values have a higher resolution than the delay values due to the rounding. In this way, oscillations of the delay values are essentially avoided.
  • the resolution of the delay values is chosen to be lower than the resolution of the delay estimated values in order to keep the technical effort involved in delaying the speech signals as low as possible.
  • the signal / noise power ratio and the speech quality of a sum signal present at the output of the adding device are improved compared to the signal / noise power ratio and the speech quality of the individual speech signals.
  • the digital filter is a digital Hilbert transformer.
  • a digital Hilbert transformer which causes a phase shift of 90 degrees for all frequencies, has the transfer function of a low pass in terms of amount, so that the rounded delay values converge well, particularly for the low frequencies that are essential for a speech signal.
  • the Hilbert transformer can also be replaced by a differentiator, for example, which also causes a phase shift of 90 degrees.
  • a differentiator has a linearly increasing transfer function in terms of amount, so that in particular the low frequencies of a speech signal are suppressed, so that there is no convergence as good as that of a Hilbert transformer.
  • means for smoothing the gradient estimates are provided.
  • the speech processing device is provided for processing three speech signals.
  • the signal / noise power ratio and the speech quality of the sum signal present at the output of the adding device can be improved in this way.
  • the invention can also be designed in such a way that the use of a linear combination of error values is provided for determining a delay estimate for the further speech signal.
  • delay means are provided for delaying the first speech signal with a fixed delay time.
  • the speech processing device is integrated in a hands-free device.
  • the speech processing device shown in FIG. 1 contains two microphones M1 and M2. These are used to convert acoustic to electrical voice signals, which are made up of speech and noise signal components.
  • the speech signal components come from a single speech source (speaker), which as a rule has different distances from the two microphones M1 and M2.
  • the speech signal components are thus highly correlated.
  • the noise signal components of the two speech signals received by the microphones M1 and M2 are ambient noises generated by the individual speech source, which can be assumed to be uncorrelated or only slightly correlated with suitable microphone spacings in the range from 10 to 60 cm if the microphones reverberated in a so-called Environment such as in the car or in an office. If the speech source and speech processing device are located in a motor vehicle, for example, the noise signal components are caused in particular by engine and driving noises.
  • the microphone signals generated by the microphones M1 and M2 are digitized by analog-digital converters 1 and 2.
  • the resulting digitized and thus present as samples x1 (i) and x2 (i) microphone signals are evaluated by a control device 3, which is used to control and set a delay element 4.
  • the sampled microphone signals x1 (i) and x2 (i) are referred to below as microphone or speech signals.
  • the delay element 4 delays the microphone signal x1 with delay values T1 that can be set by the control device 3.
  • An adding device 5 adds the microphone signal x1 (i) delayed by the delay element 4 and the microphone signal x2 (i) delayed by a delay element 16 with a constant time delay T max .
  • the delay element 16 is provided in order to be able to set both a leading and a lagging of the microphone signal x1 (i) relative to the microphone signal x2 (i).
  • a sum signal X (i) present at the output of the adding device 5 is a sampled speech signal, the signal / noise power ratio of which is increased compared to the signal / noise power ratios of the speech signals x1 (i) and x2 (i).
  • the addition by the adder 5 increases the power of the voice signal components of the two voice signals x1 (i) and x2 (i) by approximately a factor of 4 and increases the power of the noise signal components only approximately caused by a factor of 2. This results in an improvement in the power-related signal / noise power ratio of approximately 3 dB.
  • Fig. 2 the operation of the control device 3 is explained in more detail using a block diagram.
  • the speech signal estimates x1 int (i) are values that result from an interpolation of samples of the speech signal x1 (i). The determination of the speech signal estimates x1 int (i) will be explained later.
  • i is a variable which can take integer values and with which, on the one hand, sampling times of the speech signals x1 (i) and x2 (i) and, on the other hand, also program cycles of the programmable control device 3 having control means 3, are indicated, with one new sample value per speech signal in each program cycle is processed.
  • a digital filter 6 carries out a Hilbert transformation of the sample values x2 (i):
  • the digital filter 6 supplying the values x2 H (i) of x2 (i) is an FIR filter of the order K, which has coefficients h (0), h (1), ..., h (K).
  • K is sixteen, so that the digital filter 6 has seventeen coefficients.
  • the amount of the digital filter 6 has the transfer function of a low pass. It continues to produce a 90 degree phase shift.
  • the fixed phase shift of 90 degrees is the decisive property of the digital filter 6, the course of the amount of the transfer function is not decisive for the functioning of the speech processing device.
  • the digital filter 6 can thus also be implemented with the aid of a differentiator, which would, however, lead to a suppression of low-frequency components of x2 (i) and thus to a reduced performance of the speech processing device.
  • N indicates the number of samples of x1 used in the calculation. N is, for example, equal to 65.
  • the multiplication by 1 / P x2 (i) serves to avoid instabilities in the control device 3 when the delay element 4 is controlled an estimated on the short-term power P x2 (i) graded gradient degree (i) of the squares or the power of the error values e12 (i) in the program cycle i.
  • a function block 7 continuously forms estimated values SNR (i) of the associated signal / noise power ratio from the samples of the speech signal x2 (i), which are evaluated by a function block 8.
  • An evaluation of the speech signal x1 (i) instead of the speech signal x2 (i) is also possible without the functionality of the speech processing device being restricted.
  • the functioning of the function block 7 will be explained in more detail later with reference to FIGS. 6 to 8.
  • Function block 8 carries out a threshold decision regarding the estimated values SNR (i). Only if the estimated values SNR (i) lie above a predefinable threshold is an intermediate memory 9 overwritten with the newly determined gradient estimated value grad (i). This case is symbolized by the closed position of a switch 11 which is controlled by the function block 8.
  • the memory content (degree (i)) of the intermediate memory 9 is further processed by a functional unit 10.
  • a functional unit 10 In the event that an estimated value SNR (i) lies below the predefinable threshold value, the buffer 9 is not overwritten with the newly determined gradient estimated value grad (i) and it retains its old memory content, which is symbolized by the open position of the switch 11 .
  • the predefinable threshold, on which the opening and closing of the switch 11 by the function block 8 depends, is preferably between 0 and 10 dB.
  • the intermediate memory 9 supplies the gradient estimated values grad (i) stored in it to the functional unit 10, to which sample values of the speech signal x1 (i) are also supplied and which is used both for supplying the speech signal estimated values x1 int (i) and for setting the delay element 4.
  • ⁇ is a constant that has the value 0.95 in the exemplary embodiment.
  • Delay estimated values T1 '(i) are thus determined recursively.
  • ⁇ is a constant factor or convergence parameter and is in the range R x2x2 denotes an autocorrelation function of the speech signal x2 (i) at the zero position.
  • a particularly advantageous value range of ⁇ in the present exemplary embodiment is 1.5 ⁇ ⁇ 3.
  • the delay estimated values T1 '(i) can also be non-integer values, ie non-integer multiples of a sampling interval.
  • a function block 14 rounds the delay estimated values T1 '(i) to integer delay values T1 (i) with which the delay device 4 is set. The rounding operation by function block 14 is necessary since values of the speech signal x1 (i) to be delayed by the delay element 4 are only available at the corresponding sampling times.
  • Function block 15 is thus able to use the speech signal estimate x1 int (i) in program cycle i to form or interpolate a value of speech signal x1 at time i + T1 (i), ie at a time between two sampling times.
  • the described interpolation by function block 15 can be replaced by function block 15 performing low-pass filtering of the sample values x1 (i) for the interpolation of values between the sample times.
  • the corresponding true time delay between the speech signal components which is determined by the different distances from the speaker to the microphones M1 and M2, would lie between these two delay values.
  • such oscillations are avoided by using speech signal estimates x1 int (i) when the error values are formed, by which the values of the speech signal x1 (i) are also used for delays by non-integer numbers Multiples of a sampling interval are available, ie also at times other than the sampling times i of the speech signal x1 (i).
  • the function block 12 used to smooth the gradient estimated values grad (i) brings about an improved determination of the delay estimated values T1 '(i).
  • the control device 3 adapts the delay estimates T1 '(i) or the delay values T1 (i) so that the square or the power of the error values e 1 (i) is reduced from one program cycle to the next. The convergence of T1 '(i) or T1 (i) is thus ensured.
  • FIG. 3 shows a speech processing device which works in principle like the speech processing device from FIG. 1 and now has three microphones M1, M2 and M3 for the delivery of microphone or speech signals.
  • the microphone signals are fed to analog-to-digital converters 20, 21 and 22, which deliver digitized and thus sampled speech signals x1 (i), x2 (i) and x3 (i), which consist of speech and noise signal components.
  • the speech signals x1 (i) and x3 (i) are supplied to adjustable delay elements 23 and 24.
  • the speech signal x2 (i) is fed to a delay element 27 with a fixed delay time T max.
  • the output values of the delay elements 23, 24 and 27 are added to the sum signal X (i) by an adding device 25.
  • a control device 26 evaluates the sample values of the speech signals x1 (i), x2 (i) and x3 (i) and derives rounded integer delay values T1 (i) and T3 () from these sample values analogously to the mode of operation of the control device 3 from FIGS. i) ab, which correspond to the integer multiples of a sampling interval of the sampled speech signals x1 (i), x2 (i) and x3 (i) and with which the delay elements 23 and 24 are set, so that an expansion from two to three microphone to be processed or voice signals is enabled.
  • FIG. 4 shows a first embodiment of the control device 26 from FIG. 3.
  • Two functional units 10 are provided, the structure of which is identical to the structure of the functional unit 10 from FIG. 2 and which are used to set the delay elements 23 and 24 with the rounded time delay values T1 (i) and T3 (i).
  • the upper functional unit 10 provides speech signal estimates x1 int (i).
  • the lower functional unit 10 supplies speech signal estimates x3 int (i). From a difference x1 int (i) - x2 (i) and from a difference x3 int (i) - x2 (i) error values e12 (i) and e32 (i) are formed.
  • a digital filter 6 which has already been described in more detail in the explanations relating to FIG. 2, and which serves to receive the sample values x2 (i) and to supply values x2 H (i) which are obtained by a Hilbert transformation of the Samples x2 (i) are generated.
  • the values x2 H (i) are multiplied on the one hand by the error values e12 (i) and on the other hand by the error values e32 (i).
  • the first product x2 H (i) * e12 (i) is the upper, the second product x2 H (i) * e32 (i) is fed to the lower functional unit 10.
  • the arrangement of the function blocks 7 and 8, the buffer 9 and the switch 11 is carried out analogously to FIG. 2 and is not shown in FIG. 4 for reasons of clarity.
  • FIG. 5 shows a version of the control device 26 that is expanded compared to FIG. 4.
  • three digital filters 6 are now arranged instead of just one digital filter 6. These form the values x1 H (i), x2 H (i) and x3 H (i) from the speech signal samples x1 (i), x2 (i) and x3 (i) by Hilbert transformation.
  • error values e31 (i) and e32 (i) are formed in the lower half of the block diagram shown in FIG. 5.
  • the error values e31 (i) result from the difference x3 int (i) -x1 (i) .
  • the error values e32 (i) are the difference x3 int (i) -x2 (i) educated.
  • a third product 0.3 * e31 (i) * x1 H (i) and a fourth product 0.7 * e32 (i) * x2 H (i) are added up and the resulting sum is fed to the lower functional unit 10.
  • a sum signal X (i) which is improved compared to the speech processing device with two microphones according to FIG. 1 can be generated.
  • the signal / noise power ratio and thus the speech quality of the sum signal X (i) of the speech processing device according to FIG. 3 is further increased compared to the sum signal X (i) generated by the speech processing device according to FIG. 1.
  • the control device according to FIG. 5 has an increased stability compared to the control device according to FIG. 4 when used in the speech processing device according to FIG. 3.
  • the invention can be designed in such a way that the delay estimated values T1 '(i) and T3' (i) (e.g., floating point numbers) to form the delay values T1 (i) and T3 (i) are not rounded to values which correspond to an integer multiple of a sampling interval (here: integers), but to values which correspond to a multiple of a fraction of a sampling interval .
  • integers integer multiple of a sampling interval
  • rounding the delay estimated values to multiples of a value which corresponds to a quarter or half of a sampling interval is advantageous.
  • the resolution of the delay values is increased, which can thus be adjusted more precisely, so that the speech quality of the sum signals X (i) is further increased, since time differences from the speech source producing the speech signal components to the microphones M1, M2 and M3 are more precisely compensated can.
  • interpolation or low pass filtering of speech signal samples is provided to produce speech signal values that lie between two speech signal samples.
  • the interpolation or low-pass filtering can in particular be integrated into the delay means 4, 23 and 24.
  • the scheme is explained, on the basis of which the function block 7 from a sampled speech signal x (i), which consists of noise and speech signal components, the associated estimated values SNR (i) of the signal / noise power ratio, that is Ratio of the power of the speech signal components to the power of the noise signal components, determined.
  • the sample values x2 (i) correspond to the sample values x (i).
  • the function block 7 is shown in FIG. 6 on the basis of a block diagram.
  • a functional block 30 is used to form power values P x (i) of the samples x (i) by squaring the samples. Function block 30 also smoothes these power values P x (i).
  • the resulting smoothed power values P x, s (i) are supplied to both function block 31 and function block 32.
  • Function block 31 continuously determines estimated values P n (i) for estimating the power of the noise signal component of the sampled values x (i), ie the power of the noise signal components of the sampled values x (i) is determined.
  • the function block 32 continuously determines estimated values SNR (i) of the signal / noise power ratio of the sampled values x (i).
  • FIG. 7 shows a flow chart which explains the function of the function block 7 in more detail.
  • the flow chart shows how estimated values SNR (i) of the corresponding signal / noise power ratio are formed from the sampled values x (i) of the speech signal x by a computer program.
  • a counter variable Z is set to 0 and a variable P Mmin is set to a value P max .
  • P max is chosen so large that the smoothed power values P x, s (i) are always smaller than P max .
  • P max can, for example, be set to the maximum representable numerical value of a computer used to implement the program.
  • a new sample value x (i) is read in in block 34.
  • a short-term power value P x (i) of a group of N successive sample values x (i) is determined using formula (1). N here is 128, for example.
  • Equation (2) The value ⁇ from equation (2) is between 0.95 and 0.98.
  • the determination of smoothed power values P x, s (i) can also only be carried out using equation (2), in which case however the value ⁇ should be increased approximately to the value 0.99 and P x (i) by x2 (i) is replace.
  • a branch 37 queries whether the smoothed power value P x, s (i) that has just been determined is less than P Mmin . If this question is answered in the affirmative, ie P x, s (i) is less than P Mmin , block 38 sets P Mmin to the value of P x, s (i). If the question of branch 37 is answered in the negative, block 38 is skipped. This means that the minimum of M smoothed power values P x, s is in P Mmin after M program cycles . Then the branch 39 is used to query whether the counter variable Z has a value greater than or equal to a value M. In this way it is determined whether M smoothed power values have already been processed.
  • SNR (i) [P x, p (i) - min ⁇ c * P n (i), P x, p (i) ⁇ ] / [c * P n (i)] a current estimate SNR (i) of the signal / noise power ratio of the speech signal x (i) is determined.
  • the product c * P n (i) is used to estimate the current power of the noise signal component
  • the difference P x, p (i) -c * P n (i) is used to estimate the current power of the speech signal component of the speech signal x (i).
  • the current power of the speech signal is estimated by the smoothed power value P x, s (i).
  • the weighting with a scaling factor c prevents P n (i) from estimating the noise signal power with a value that is too small.
  • the scaling factor c is typically in the range from 1.3 to 2.
  • the minimum formation in block 41 or equation (4) ensures that the non-logarithmic signal / noise power ratio SNR (i) is also positive if in exceptional cases c * P n (i) is greater than P x, s (i). Then the power of the noise signal component of the voice signal is set equal to the power of the voice signal estimated by P x, s (i).
  • the power of the speech signal component of the speech signal estimated by P x, s (i) -P x, s (i) is then equal to zero, as is the non-logarithmic signal / noise power ratio.
  • the program continues with the reading in of a new speech signal sample value x (i) by block 34.
  • P n (i) is set equal to P Mmin in block 45, so that an adaptation of the estimation of the noise signal component is accelerated takes place since P n (i) is determined at the minimum of the last (M ⁇ L) values. Then in block 46 the counter variable Z is reset to 0 and P Mmin again receives the value P max .
  • M successive smoothed P x, s (i) samples x (i) of the speech signal x are combined into a subgroup.
  • the minimum of the smoothed power values P x, s (i) is determined by the operations carried out with branch 37 and block 38.
  • the W minima determined last are stored in the components of the vector minvec. If the last W minima are not monotonically increasing (see branch 43), then a preliminary estimate P n (i) of the power of the noise signal component is determined from the minimum of the minima of the last W subgroups, ie from the minimum of a group, according to block 44.
  • the minimum of the last subgroup with M smoothed power values P x is determined by block 45 to estimate the current estimated value P n (i) of the power of the noise signal component . s (i) used. This shortens the time period with which monotonically increasing smoothed power values P x, s (i) also cause a change in the estimated values SNR (i).
  • the value P n (i) is determined from the minimum of the last W subgroup minima or the last L smoothed power values P x, s (i), which is used to estimate the noise signal power.
  • a realization with adjacent, ie non-overlapping groups is also conceivable. However, the time span between two estimated values SNR (i) is then increased with a reduced computing effort, so that the reaction time to changing SNR of the speech signal x (i) is increased.
  • the described speech processing device thus has an estimation device which is suitable for the continuous formation of estimated values SNR (i) of the signal / noise power ratio of noisy speech signals x (i). In particular, no speech pauses are required to estimate the noise signal power.
  • the estimation device described uses the special time profile of smoothed power values of the speech signal x (i), which is characterized by peaks and intermediate areas with smaller smoothed power values P x, s (i), their temporal expansion from the respective speech source, ie the respective speaker , depends. The areas between the peaks are used to estimate the power of the noise signal component.
  • the groups with L smoothed power values P x, s (i) must follow one another without gaps, ie they must either adjoin or overlap.
  • each group must contain so many smoothed power values P x, s (i) that at least all values belonging to any peak can be recorded. Since the most extended peaks can be estimated by the most extended phonemes of a speech signal, ie the vowels, the number L describing the group size can be derived from this. For a sampling rate of the speech signal of 8 kHz, a useful value of L is in the range between 3000 and 8000. An advantageous value for W is 4. With such a dimensioning, there is a good compromise between the computational effort and the speed of reaction of the function block 7.
  • FIG. 9 shows a use of the voice processing device from FIG. 3 in a mobile radio terminal 50.
  • the language processing means 20 to 26 are combined in a function block 51, which forms the sum signal values X (i) from the microphone or speech signals generated by the microphones M1, M2 and M3.
  • the microphones M1, M2 and M3 advantageously have a distance of 10 to 60 cm, so that in a so-called "reverberated" environment (eg car, office) the interference signal components of the speech signals supplied by the microphones M1, M2 and M3 are largely uncorrelated. This also applies when only two microphones are used as in FIG. 1.
  • a function block 52 which processes the sum signal values X (i) combines all the other means of the mobile radio terminal 50 for receiving, processing and transmitting signals which are used for communication with a base station (not shown) serve, the transmission and reception of signals via an antenna 54 coupled to the function block 52. Furthermore, a loudspeaker 53 coupled to the function block 52 is provided. A user (speaker, listener) communicates acoustically with the mobile radio terminal 50 via the microphones M1 to M3 and the loudspeaker 53, which are part of a hands-free device integrated in the mobile radio terminal 50.
  • the use of such a mobile radio terminal 50 is particularly advantageous in motor vehicles, since there the hands-free communication via the mobile radio terminal is particularly disturbed by engine or driving noise (noise).

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Noise Elimination (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP95201578A 1994-06-22 1995-06-14 Dispositif pour le traitement de paroles et émetteur-récepteur mobile Expired - Lifetime EP0689191B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE4421853A DE4421853A1 (de) 1994-06-22 1994-06-22 Mobilfunkendgerät
DE4421853 1994-06-22

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EP0689191A2 true EP0689191A2 (fr) 1995-12-27
EP0689191A3 EP0689191A3 (fr) 1997-05-28
EP0689191B1 EP0689191B1 (fr) 2001-05-23

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US (1) US5647006A (fr)
EP (1) EP0689191B1 (fr)
JP (1) JPH0818473A (fr)
DE (2) DE4421853A1 (fr)

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JP5931108B2 (ja) * 2014-03-20 2016-06-08 本田技研工業株式会社 ナビゲーションサーバ及びプログラム

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DE59509271D1 (de) 2001-06-28
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JPH0818473A (ja) 1996-01-19
EP0689191B1 (fr) 2001-05-23
DE4421853A1 (de) 1996-01-04

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